diff --git a/webrtc/WebRtcTransport.cpp b/webrtc/WebRtcTransport.cpp index ccdeab38..98aa3ee5 100644 --- a/webrtc/WebRtcTransport.cpp +++ b/webrtc/WebRtcTransport.cpp @@ -255,7 +255,8 @@ void WebRtcTransport::inputSockData(char *buf, size_t len, RTC::TransportTuple * if (_srtp_session_recv->DecryptSrtp((uint8_t *) buf, &len)) { onRtp(buf, len); } else { - WarnL; + RtpHeader *rtp = (RtpHeader *) buf; + WarnL << "srtp_unprotect rtp failed, pt:" << (int)rtp->pt; } return; } @@ -411,12 +412,10 @@ void WebRtcTransportImp::onStartWebRTC() { info->rtcp_context_send = std::make_shared(info->plan_rtp->sample_rate, false); //send ssrc --> RtpPayloadInfo - _rtp_info_ssrc[info->answer_ssrc_rtp] = std::make_pair(false, info); - _rtp_info_ssrc[info->answer_ssrc_rtx] = std::make_pair(true, info); + _rtp_info_ssrc[info->answer_ssrc_rtp] = info; //recv ssrc --> RtpPayloadInfo - _rtp_info_ssrc[info->offer_ssrc_rtp] = std::make_pair(false, info);; - _rtp_info_ssrc[info->offer_ssrc_rtx] = std::make_pair(true, info);; + _rtp_info_ssrc[info->offer_ssrc_rtp] = info; //rtp pt --> RtpPayloadInfo _rtp_info_pt.emplace(info->plan_rtp->pt, std::make_pair(false, info)); @@ -589,17 +588,14 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { RtcpSR *sr = (RtcpSR *) rtcp; auto it = _rtp_info_ssrc.find(sr->ssrc); if (it != _rtp_info_ssrc.end()) { - auto rtx = it->second.first; - if (!rtx) { - auto &info = it->second.second; - auto it = info->rtcp_context_recv.find(sr->ssrc); - if (it != info->rtcp_context_recv.end()) { - it->second->onRtcp(sr); - auto rr = it->second->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc); - sendRtcpPacket(rr->data(), rr->size(), true); - } else { - WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); - } + auto &info = it->second; + auto it = info->rtcp_context_recv.find(sr->ssrc); + if (it != info->rtcp_context_recv.end()) { + it->second->onRtcp(sr); + auto rr = it->second->createRtcpRR(info->answer_ssrc_rtp, sr->ssrc); + sendRtcpPacket(rr->data(), rr->size(), true); + } else { + WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); } } else { WarnL << "未识别的sr rtcp包:" << rtcp->dumpString(); @@ -613,12 +609,9 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { for (auto item : rr->getItemList()) { auto it = _rtp_info_ssrc.find(item->ssrc); if (it != _rtp_info_ssrc.end()) { - auto rtx = it->second.first; - if (!rtx) { - auto &info = it->second.second; - auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp); - sendRtcpPacket(sr->data(), sr->size(), true); - } + auto &info = it->second; + auto sr = info->rtcp_context_send->createRtcpSR(info->answer_ssrc_rtp); + sendRtcpPacket(sr->data(), sr->size(), true); } else { WarnL << "未识别的rr rtcp包:" << rtcp->dumpString(); } @@ -653,15 +646,12 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { WarnL << "未识别的 rtcp包:" << rtcp->dumpString(); return; } - auto rtx = it->second.first; - if (!rtx) { - auto &info = it->second.second; - auto &fci = fb->getFci(); - info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { - //rtp重传 - onSendRtp(rtp, true, true); - }); - } + auto &info = it->second; + auto &fci = fb->getFci(); + info->nack_list.for_each_nack(fci, [&](const RtpPacket::Ptr &rtp) { + //rtp重传 + onSendRtp(rtp, true, true); + }); break; } default: break; @@ -675,7 +665,8 @@ void WebRtcTransportImp::onRtcp(const char *buf, size_t len) { /////////////////////////////////////////////////////////////////// -void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx, string *rid_ptr) const{ +void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr) const{ + string rid, repaired_rid; auto ext_map = RtpExt::getExtValue(header); for (auto &pr : ext_map) { if (is_recv) { @@ -688,33 +679,9 @@ void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *h pr.second.setType(it->second); //重新赋值ext id为 ext type，作为后面处理ext的统一中间类型 pr.second.setExtId((uint8_t) it->second); - switch(it->second){ - case RtpExtType::sdes_repaired_rtp_stream_id : - case RtpExtType::sdes_rtp_stream_id : { - auto ssrc = ntohl(header->ssrc); - auto rid = it->second == RtpExtType::sdes_rtp_stream_id ? pr.second.getRtpStreamId() : pr.second.getRepairedRtpStreamId(); - //根据rid获取rtp或rtx的ssrc - auto &ssrc_ref = is_rtx ? info.rid_ssrc[rid].second : info.rid_ssrc[rid].first; - if (!ssrc_ref) { - //ssrc未赋值，赋值 - ssrc_ref = ssrc; - DebugL << (is_rtx ? "got rid of rtx:" : "got rid:") << rid << ", ssrc:" << ssrc; - } - if (is_rtx) { - //rtx ssrc --> rtp ssrc - auto &rtp_ssrc_ref = info.rtx_ssrc_to_rtp_ssrc[ssrc]; - if (!rtp_ssrc_ref && info.rid_ssrc[rid].first) { - //未找到rtx到rtp ssrc的映射关系，且已经获取rtp的ssrc，那么设置映射关系 - rtp_ssrc_ref = info.rid_ssrc[rid].first; - DebugL << "got ssrc of rid:" << rid << ", [rtx-rtp]:" << ssrc << "-" << rtp_ssrc_ref; - } - } - if (rid_ptr) { - *rid_ptr = rid; - } - break; - } - + switch (it->second) { + case RtpExtType::sdes_rtp_stream_id : rid = pr.second.getRtpStreamId(); break; + case RtpExtType::sdes_repaired_rtp_stream_id : repaired_rid = pr.second.getRepairedRtpStreamId(); break; default : break; } } else { @@ -729,74 +696,98 @@ void WebRtcTransportImp::changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *h pr.second.setExtId(it->second); } } + + if (!is_recv) { + return; + } + if (rid.empty()) { + rid = repaired_rid; + } + auto ssrc = ntohl(header->ssrc); + if (rid.empty()) { + //获取rid + rid = info.ssrc_to_rid[ssrc]; + } else { + //设置rid + info.ssrc_to_rid[ssrc] = rid; + } + if (rid_ptr) { + *rid_ptr = rid; + } +} + +std::shared_ptr WebRtcTransportImp::createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info){ + auto ref = std::make_shared([info, this, rid](RtpPacket::Ptr rtp) mutable { + onSortedRtp(*info, rid, std::move(rtp)); + }); + + if (!is_rtx) { + //rtx没nack + info->nack_ctx[rid].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable { + onSendNack(*info, nack, ssrc); + }); + //ssrc --> RtpPayloadInfo + _rtp_info_ssrc[ssrc] = info; + } + + InfoL << "receive rtp of ssrc:" << ssrc << ", rid:" << rid << ", is rtx:" << is_rtx << ", codec:" << info->plan_rtp->codec; + return ref; } void WebRtcTransportImp::onRtp(const char *buf, size_t len) { - onRtp_l(buf, len, false); -} - -void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { - if (!rtx) { - _bytes_usage += len; - _alive_ticker.resetTime(); - } + _bytes_usage += len; + _alive_ticker.resetTime(); RtpHeader *rtp = (RtpHeader *) buf; - auto ssrc = ntohl(rtp->ssrc); - //根据接收到的rtp的pt信息，找到该流的信息 auto it = _rtp_info_pt.find(rtp->pt); if (it == _rtp_info_pt.end()) { - WarnL; + WarnL << "unknown rtp pt:" << (int)rtp->pt; return; } + bool is_rtx = it->second.first; + auto ssrc = ntohl(rtp->ssrc); auto &info = it->second.second; - if (!it->second.first) { + + //修改ext id至统一 + string rid; + changeRtpExtId(*info, rtp, true, &rid); + +#if 0 + if (rid.empty() && info->media->type == TrackVideo) { + WarnL << "ssrc:" << ssrc << ", rtx:" << is_rtx << ",seq:" << ntohs((uint16_t) rtp->seq); + } +#endif + auto &ref = info->receiver[rid]; + if (!ref) { + ref = createRtpReceiver(rid, ssrc, is_rtx, info); + } + + if (!is_rtx) { //这是普通的rtp数据 auto seq = ntohs(rtp->seq); #if 0 - if (!rtx && info->media->type == TrackVideo && seq % 100 == 0) { + if (info->media->type == TrackVideo && seq % 100 == 0) { //此处模拟接受丢包 DebugL << "recv dropped:" << seq; return; } #endif - auto &ref = info->receiver[ssrc]; - if (!rtx) { - //统计rtp接受情况，便于生成nack rtcp包 - info->nack_ctx[ssrc].received(seq); - //时间戳转换成毫秒 - auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate; + //统计rtp接受情况，便于生成nack rtcp包 + info->nack_ctx[rid].received(seq); - //统计rtp收到的情况，好做rr汇报 - auto &cxt_ref = info->rtcp_context_recv[ssrc]; - if (!cxt_ref) { - cxt_ref = std::make_shared(info->plan_rtp->sample_rate, true); - } - cxt_ref->onRtp(seq, stamp_ms, len); - - //修改ext id至统一 - string rid; - changeRtpExtId(*info, rtp, true, false, &rid); - - if (!ref) { - ref = std::make_shared([info, this, rid](RtpPacket::Ptr rtp) mutable { - onSortedRtp(*info, rid, std::move(rtp)); - }); - info->nack_ctx[ssrc].setOnNack([info, this, ssrc](const FCI_NACK &nack) mutable { - onSendNack(*info, nack, ssrc); - }); - //recv simulcast ssrc --> RtpPayloadInfo - _rtp_info_ssrc[ssrc] = std::make_pair(false, info); - InfoL << "receive rtp of ssrc:" << ssrc; - } + auto &cxt_ref = info->rtcp_context_recv[ssrc]; + if (!cxt_ref) { + cxt_ref = std::make_shared(info->plan_rtp->sample_rate, true); + info->rid_to_ssrc[rid] = ssrc; } + //时间戳转换成毫秒 + auto stamp_ms = ntohl(rtp->stamp) * uint64_t(1000) / info->plan_rtp->sample_rate; + //统计rtp收到的情况，好做rr汇报 + cxt_ref->onRtp(seq, stamp_ms, len); + //解析并排序rtp - if (ref) { - ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); - } else { - WarnL << "rtp dropped, ssrc:" << ssrc << ", is rtx:" << rtx; - } + ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); return; } @@ -808,26 +799,18 @@ void WebRtcTransportImp::onRtp_l(const char *buf, size_t len, bool rtx) { return; } - //修改ext id至统一 - changeRtpExtId(*info, rtp, true, true); - //前两个字节是原始的rtp的seq auto origin_seq = payload[0] << 8 | payload[1]; + //rtx的seq转换为rtp的seq rtp->seq = htons(origin_seq); - if (info->offer_ssrc_rtp) { - //非simulcast或音频 - rtp->ssrc = htonl(info->offer_ssrc_rtp); - TraceL << "received rtx rtp,ssrc: " << ssrc << ", seq:" << origin_seq << ", pt:" << (int)rtp->pt; - } else { - //todo simulcast下，辅码流通过rtx传输？ - //simulcast情况下，根据rtx的ssrc查找rtp的ssrc - rtp->ssrc = htonl(info->rtx_ssrc_to_rtp_ssrc[ntohl(rtp->ssrc)]); - } + //rtx的ssrc转换为rtp的ssrc + rtp->ssrc = htonl(info->rid_to_ssrc[rid]); + //rtx的pt转换为rtp的pt rtp->pt = info->plan_rtp->pt; memmove((uint8_t *) buf + 2, buf, payload - (uint8_t *) buf); buf += 2; len -= 2; - onRtp_l(buf, len, true); + ref->inputRtp(info->media->type, info->plan_rtp->sample_rate, (uint8_t *) buf, len); } void WebRtcTransportImp::onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc) { @@ -908,12 +891,12 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, size_t &len, void * if (!pr->first || !pr->second->plan_rtx) { //普通的rtp,或者不支持rtx, 修改目标pt和ssrc - changeRtpExtId(*pr->second, header, false, false); + changeRtpExtId(*pr->second, header, false); header->pt = pr->second->plan_rtp->pt; header->ssrc = htonl(pr->second->answer_ssrc_rtp); } else { //重传的rtp, rtx - changeRtpExtId(*pr->second, header, false, true); + changeRtpExtId(*pr->second, header, false); header->pt = pr->second->plan_rtx->pt; if (pr->second->answer_ssrc_rtx) { //有rtx单独的ssrc,有些情况下，浏览器支持rtx，但是未指定rtx单独的ssrc diff --git a/webrtc/WebRtcTransport.h b/webrtc/WebRtcTransport.h index c62f549e..ef863fcd 100644 --- a/webrtc/WebRtcTransport.h +++ b/webrtc/WebRtcTransport.h @@ -298,8 +298,6 @@ protected: void onRtcConfigure(RtcConfigure &configure) const override; void onRtp(const char *buf, size_t len) override; - void onRtp_l(const char *buf, size_t len, bool rtx); - void onRtcp(const char *buf, size_t len) override; void onBeforeEncryptRtp(const char *buf, size_t &len, void *ctx) override; void onBeforeEncryptRtcp(const char *buf, size_t &len, void *ctx) override {}; @@ -351,16 +349,19 @@ private: const RtcMedia *media; NackList nack_list; RtcpContext::Ptr rtcp_context_send; - unordered_map > rid_ssrc; - unordered_map rtx_ssrc_to_rtp_ssrc; - unordered_map nack_ctx; + //只生成rtp的nack，rtx的不生成 + unordered_map nack_ctx; + unordered_map rid_to_ssrc; + unordered_map > receiver; + unordered_map ssrc_to_rid; + //只统计rtp的接收情况，rtx的不统计 unordered_map rtcp_context_recv; - unordered_map > receiver; }; void onSortedRtp(RtpPayloadInfo &info, const string &rid, RtpPacket::Ptr rtp); void onSendNack(RtpPayloadInfo &info, const FCI_NACK &nack, uint32_t ssrc); - void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, bool is_rtx = false, string *rid_ptr = nullptr) const; + void changeRtpExtId(RtpPayloadInfo &info, const RtpHeader *header, bool is_recv, string *rid_ptr = nullptr) const; + std::shared_ptr createRtpReceiver(const string &rid, uint32_t ssrc, bool is_rtx, const RtpPayloadInfo::Ptr &info); private: uint16_t _rtx_seq[2] = {0, 0}; @@ -389,8 +390,8 @@ private: RtpPayloadInfo::Ptr _send_rtp_info[2]; //根据接收rtp的pt获取相关信息 unordered_map > _rtp_info_pt; - //根据rtcp的ssrc获取相关信息 - unordered_map > _rtp_info_ssrc; + //根据rtcp的ssrc获取相关信息，只记录rtp的ssrc，rtx的ssrc不记录 + unordered_map _rtp_info_ssrc; //发送rtp时需要修改rtp ext id map _rtp_ext_type_to_id; //接收rtp时需要修改rtp ext id